Over the past several years there has been, and continues to be, a tremendous amount of activity in the area of efficient encoding of speech. A large number of digital coding algorithms are being investigated for a wide variety of applications. For an evolving digital telephone network, a most important application is the possible replacement of 64,000 bit-per-second (bps) PCM signal (8 bits per time slot, repeated at an 8 kHz rate) for telephony--both in the public switched and private line networks. The reasons, of course, is to achieve bandwidth compression. The dilemma for telephony planners is easily posed but not so easily answered--should such a network evolve toward a coding elgorithm more efficient than 64 kb/s PCM and, if so, which algorithm is preferable? A number of different digital coding algorithms and related techniques have been proposed heretofore, namely: Adaptive Differential PCM (ADPCM); Sub-Band Coding (SBC); Time Domain Harmonic Scaling (TDHS); vocoder-driven Adaptive Transform Coding (ATC), etc.
For a realistic mix of input speech and voiceband data, the ADPCM approach appears to be the most promising. Adaptive differential PCM coding is disclosed, for example, in the article by Cummiskey-Jayant-Flanagan (CJF) entitled "Adaptive Quantization in Differential PCM Coding of Speech," Bell System Technical Journal, Vol. 52, No. 7, September 1973, pp. 1105-1118. The performance of the CJF-ADPCM algorithm has been well established in previous studies (W. R. Daumer, J. R. Cavanaugh, "A Subjective Comparison of Selected Digital Codecs for Speech", Bell System Technical Journal, Vol. 57, No. 9, November 1978, pp. 3119-3165) and can be considered a benchmark since a single encoding with this coder at 32 kb/s is near to being subjectively equivalent to 64 kg/s .mu.to 255 PCM. However, it is not as robust in a tandem encoding situation (see the above referenced Daumer-Cavanaugh article) and it will not handle higher speed voiceband data (e.g., 4800 bps) in a realistic mixed analog/digital network.